1. Field
The following description generally relates to encoders and decoders and, in particular, to an efficient MDCT/IMDCT implementation for voice and audio codecs.
2. Background
One goal of audio coding is to compress an audio signal into a desired limited information quantity while keeping as much as the original sound quality as possible. In an encoding process, an audio signal in a time domain is transformed into a frequency domain, and a corresponding decoding process reverses such operation.
As part of such an encoding process, a signal may be processed by a modified discrete cosine transform (MDCT). The modified discrete cosine transform (MDCT) is a Fourier-related transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property that blocks are overlapped so that the ending of one block coincides with the beginning of the next block. This overlapping helps to avoid aliasing artifacts, and in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications.
MDCT transform has also found applications in speech compression. ITU-T G.722.1 and G.722.1C vocoders apply MDCT on input speech signal, while more recent ITU-T G.729.1 and G.718 algorithms use it to process residual signal, remaining after the use of Code Excited Linear Prediction (CELP) encoder. The above mentioned vocoders operate with input sampling rates of either 8 kHz or 16 kHz, and 10 or 20-millisecond frames. Hence, their MDCT filterbanks are either 160 or 320-point transforms.
However, if future speech coders will support block-switching functionality support for decimated sizes (160, 80, 40-points) may also be needed.